1. Field of the Invention
The present invention relates to an analog filter and an audio amplifier using the analog filter. The filter is used in a digital audio system to reproduce original audio signals having a basic frequency component and to eliminate unnecessary harmonic components.
2. Discussion of the Related Art
In a digital audio system, an analog to digital converter (hereinafter referred to as an A/D converter) is used at the first stage of a digital audio system, where the audio signal is changed to a digital signal. A digital to analog converter (hereinafter refers to as a D/A converter) is used at the final output stage in order to convert the digital signal to an analog signal. The output signal from the D/A converter contains higher frequency harmonic components than the output signal from the A/D converter. The output signal contains a basic frequency component up to a frequency of 20 kHz, as well as upper and lower side band components of the integral multiples of the sampling frequency fs. In a digital audio system, a low pass filter is indispensable to eliminate frequency components higher than the sampling frequency and to reproduce its basic component up to 20 kHz.
The most common music medium for recording is the compact disc. In a compact disc system, the upper limit of the basic frequency is 20 kHz and the sampling frequency is 44.1 kHz. Thus, there are unnecessary side band components close to the upper limit of the basic frequency. If these unnecessary components are reproduced by a speaker system, cross modulation distortion may occur, or a tweeter may be damaged. In addition, some people may hear these unnecessary side band components in the music. The low pass filter is used to eliminate such unnecessary components higher than 20 kHz.
However, the lowest frequency of the unnecessary side band, 24.1 kHz, is close to the highest frequency of the basic component. A low pass filter having very sharp attenuation can be used to eliminate unnecessary frequency components. For example, some compact disc players have an analog low pass filter of a high order, such as a Tchebyscheff-type low pass filter of the 11th order.
FIG. 1A shows an example of a circuit diagram of the Tchebyscheff-type low pass filter 100 of the 11th order, whose cutoff frequency is 20 kHz. FIG. 2 shows the amplitude and phase characteristics of the low pass filter. As shown in this Figure, the amplitude characteristic represented by the dotted line is almost flat up to around 20 kHz. However, as shown by the solid line, the phase shift gradually increases within the range of 200 Hz to 9 kHz, and fluctuates dramatically at a frequency of 9 kHz and higher. This phase shift can cause serious negative effects in the high fidelity audio equipment.
In an attempt to solve this problem, an over sampling technique is used. In over sampling, a digital filter is installed in front of a D/A converter, where the frequency component that is higher than the basic frequency is cut off by the digital filter. Then the digital signal is converted to an analog signal at a frequency of integral multiples of the sampling frequency. Thereafter, an analog low pass filter of a lower order, such as a second or third order, attenuates harmonic components of higher frequency. FIG. 1B shows a low pass filter 110 of the third order which is used at the output stage of a four times over sampling circuit having a cutoff frequency of 80 kHz.
Further, FIG. 3 shows the characteristics of the low pass filter. As shown in FIG. 3, the phase shift gradually increases and reaches 30xc2x0 at 20 kHz. Such a phase shift can cause negative influences on high fidelity audio equipment.
It is very important to improve not only the frequency characteristic but also the phase characteristic to reproduce a high fidelity audio signal. If the phase characteristic is poor, high fidelity sound cannot be obtained. These problems have not yet been solved in any conventional digital audio system.
The present invention solves the problems associated with low pass filters connected to a D/A converter. Thus, an object of the invention is to provide an analog filter that has an improved phase characteristic as well as an improved amplitude characteristic within the basic frequency.
The analog filter of the present invention can be connected to a digital to analog converter, which converts digital signals sampled, at a specified sampling frequency fs, to an analog signal for a digital audio system. The filter comprises a number n of band elimination filters having cutoff frequencies of fsxc2x1xcex94f, 2fsxc2x1xcex94f, 3fsxc2x1xcex94f, nfsxc2x1xcex94f, respectively, where n is a natural number.